/*
**
** Copyright 2007, The Android Open Source Project
**
** Licensed under the Apache License, Version 2.0 (the "License");
** you may not use this file except in compliance with the License.
** You may obtain a copy of the License at
**
**     http://www.apache.org/licenses/LICENSE-2.0
**
** Unless required by applicable law or agreed to in writing, software
** distributed under the License is distributed on an "AS IS" BASIS,
** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
** See the License for the specific language governing permissions and
** limitations under the License.
*/

#pragma once

#include <stdint.h>
#include <sys/types.h>
#include <pthread.h>

#include "audio/android/AudioBufferProvider.h"
#include "audio/android/AudioResamplerPublic.h"

#include "audio/android/AudioResampler.h"
#include "audio/android/audio.h"

// IDEA: This is actually unity gain, which might not be max in future, expressed in U.12
#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT

namespace cc {

// ----------------------------------------------------------------------------

class AudioMixer {
public:
    AudioMixer(size_t frameCount, uint32_t sampleRate,
               uint32_t maxNumTracks = MAX_NUM_TRACKS);

    /*virtual*/ ~AudioMixer(); // non-virtual saves a v-table, restore if sub-classed

    // This mixer has a hard-coded upper limit of 32 active track inputs.
    // Adding support for > 32 tracks would require more than simply changing this value.
    static const uint32_t MAX_NUM_TRACKS = 32;
    // maximum number of channels supported by the mixer

    // This mixer has a hard-coded upper limit of 8 channels for output.
    static const uint32_t MAX_NUM_CHANNELS = 8;
    static const uint32_t MAX_NUM_VOLUMES = 2; // stereo volume only
    // maximum number of channels supported for the content
    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;

    static const uint16_t UNITY_GAIN_INT = 0x1000;
    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;

    enum { // names

        // track names (MAX_NUM_TRACKS units)
        TRACK0 = 0x1000,

        // 0x2000 is unused

        // setParameter targets
        TRACK = 0x3000,
        RESAMPLE = 0x3001,
        RAMP_VOLUME = 0x3002, // ramp to new volume
        VOLUME = 0x3003,      // don't ramp
        TIMESTRETCH = 0x3004,

        // set Parameter names
        // for target TRACK
        CHANNEL_MASK = 0x4000,
        FORMAT = 0x4001,
        MAIN_BUFFER = 0x4002,
        AUX_BUFFER = 0x4003,
        DOWNMIX_TYPE = 0X4004,
        MIXER_FORMAT = 0x4005,       // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
        // for target RESAMPLE
        SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
                              // parameter 'value' is the new sample rate in Hz.
                              // Only creates a sample rate converter the first time that
                              // the track sample rate is different from the mix sample rate.
                              // If the new sample rate is the same as the mix sample rate,
                              // and a sample rate converter already exists,
                              // then the sample rate converter remains present but is a no-op.
        RESET = 0x4101,       // Reset sample rate converter without changing sample rate.
                              // This clears out the resampler's input buffer.
        REMOVE = 0x4102,      // Remove the sample rate converter on this track name;
                              // the track is restored to the mix sample rate.
        // for target RAMP_VOLUME and VOLUME (8 channels max)
        // IDEA: use float for these 3 to improve the dynamic range
        VOLUME0 = 0x4200,
        VOLUME1 = 0x4201,
        AUXLEVEL = 0x4210,
        // for target TIMESTRETCH
        PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
                                // parameter 'value' is a pointer to the new playback rate.
    };

    // For all APIs with "name": TRACK0 <= name < TRACK0 + MAX_NUM_TRACKS

    // Allocate a track name.  Returns new track name if successful, -1 on failure.
    // The failure could be because of an invalid channelMask or format, or that
    // the track capacity of the mixer is exceeded.
    int getTrackName(audio_channel_mask_t channelMask,
                     audio_format_t format, int sessionId);

    // Free an allocated track by name
    void deleteTrackName(int name);

    // Enable or disable an allocated track by name
    void enable(int name);
    void disable(int name);

    void setParameter(int name, int target, int param, void *value);

    void setBufferProvider(int name, AudioBufferProvider *bufferProvider);
    void process(int64_t pts);

    uint32_t trackNames() const { return mTrackNames; }

    size_t getUnreleasedFrames(int name) const;

    static inline bool isValidPcmTrackFormat(audio_format_t format) {
        switch (format) {
            case AUDIO_FORMAT_PCM_8_BIT:
            case AUDIO_FORMAT_PCM_16_BIT:
            case AUDIO_FORMAT_PCM_24_BIT_PACKED:
            case AUDIO_FORMAT_PCM_32_BIT:
            case AUDIO_FORMAT_PCM_FLOAT:
                return true;
            default:
                return false;
        }
    }

private:
    enum {
        // IDEA: this representation permits up to 8 channels
        NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
    };

    enum {
        NEEDS_CHANNEL_1 = 0x00000000, // mono
        NEEDS_CHANNEL_2 = 0x00000001, // stereo

        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT

        NEEDS_MUTE = 0x00000100,
        NEEDS_RESAMPLE = 0x00001000,
        NEEDS_AUX = 0x00010000,
    };

    struct state_t;
    struct track_t;

    typedef void (*hook_t)(track_t *t, int32_t *output, size_t numOutFrames, int32_t *temp,
                           int32_t *aux);
    static const int BLOCKSIZE = 16; // 4 cache lines

    struct track_t {
        uint32_t needs;

        // REFINE: Eventually remove legacy integer volume settings
        union {
            int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
            int32_t volumeRL;
        };

        int32_t prevVolume[MAX_NUM_VOLUMES];

        // 16-byte boundary

        int32_t volumeInc[MAX_NUM_VOLUMES];
        int32_t auxInc;
        int32_t prevAuxLevel;

        // 16-byte boundary

        int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
        uint16_t frameCount;

        uint8_t channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
        uint8_t unused_padding; // formerly format, was always 16
        uint16_t enabled;       // actually bool
        audio_channel_mask_t channelMask;

        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
        AudioBufferProvider *bufferProvider;

        // 16-byte boundary

        mutable AudioBufferProvider::Buffer buffer; // 8 bytes

        hook_t hook;
        const void *in; // current location in buffer

        // 16-byte boundary

        AudioResampler *resampler;
        uint32_t sampleRate;
        int32_t *mainBuffer;
        int32_t *auxBuffer;

        // 16-byte boundary

        /* Buffer providers are constructed to translate the track input data as needed.
         *
         * REFINE: perhaps make a single PlaybackConverterProvider class to move
         * all pre-mixer track buffer conversions outside the AudioMixer class.
         *
         * 1) mInputBufferProvider: The AudioTrack buffer provider.
         * 2) mReformatBufferProvider: If not NULL, performs the audio reformat to
         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
         *    requires reformat. For example, it may convert floating point input to
         *    PCM_16_bit if that's required by the downmixer.
         * 3) downmixerBufferProvider: If not NULL, performs the channel remixing to match
         *    the number of channels required by the mixer sink.
         * 4) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
         *    the downmixer requirements to the mixer engine input requirements.
         * 5) mTimestretchBufferProvider: Adds timestretching for playback rate
         */
        AudioBufferProvider *mInputBufferProvider; // externally provided buffer provider.
                                                   //cjh        PassthruBufferProvider*  mReformatBufferProvider; // provider wrapper for reformatting.
                                                   //        PassthruBufferProvider*  downmixerBufferProvider; // wrapper for channel conversion.
                                                   //        PassthruBufferProvider*  mPostDownmixReformatBufferProvider;
                                                   //        PassthruBufferProvider*  mTimestretchBufferProvider;

        int32_t sessionId;

        audio_format_t mMixerFormat;           // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
        audio_format_t mFormat;                // input track format
        audio_format_t mMixerInFormat;         // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
                                               // each track must be converted to this format.
        audio_format_t mDownmixRequiresFormat; // required downmixer format
                                               // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
                                               // AUDIO_FORMAT_INVALID if no required format

        float mVolume[MAX_NUM_VOLUMES];     // floating point set volume
        float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
        float mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment

        float mAuxLevel;     // floating point set aux level
        float mPrevAuxLevel; // floating point prev aux level
        float mAuxInc;       // floating point aux increment

        audio_channel_mask_t mMixerChannelMask;
        uint32_t mMixerChannelCount;

        AudioPlaybackRate mPlaybackRate;

        bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
        bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
        bool doesResample() const { return resampler != NULL; }
        void resetResampler() {
            if (resampler != NULL) resampler->reset();
        }
        void adjustVolumeRamp(bool aux, bool useFloat = false);
        size_t getUnreleasedFrames() const { return resampler != NULL ? resampler->getUnreleasedFrames() : 0; };

        status_t prepareForDownmix();
        void unprepareForDownmix();
        status_t prepareForReformat();
        void unprepareForReformat();
        bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
        void reconfigureBufferProviders();
    };

    typedef void (*process_hook_t)(state_t *state, int64_t pts);

    // pad to 32-bytes to fill cache line
    struct state_t {
        uint32_t enabledTracks;
        uint32_t needsChanged;
        size_t frameCount;
        process_hook_t hook; // one of process__*, never NULL
        int32_t *outputTemp;
        int32_t *resampleTemp;
        //cjh        NBLog::Writer*  mLog;
        int32_t reserved[1];
        // IDEA: allocate dynamically to save some memory when maxNumTracks < MAX_NUM_TRACKS
        track_t tracks[MAX_NUM_TRACKS] __attribute__((aligned(32)));
    };

    // bitmask of allocated track names, where bit 0 corresponds to TRACK0 etc.
    uint32_t mTrackNames;

    // bitmask of configured track names; ~0 if maxNumTracks == MAX_NUM_TRACKS,
    // but will have fewer bits set if maxNumTracks < MAX_NUM_TRACKS
    const uint32_t mConfiguredNames;

    const uint32_t mSampleRate;

    //cjh    NBLog::Writer   mDummyLog;
public:
    //cjh    void            setLog(NBLog::Writer* log);
private:
    state_t mState __attribute__((aligned(32)));

    // Call after changing either the enabled status of a track, or parameters of an enabled track.
    // OK to call more often than that, but unnecessary.
    void invalidateState(uint32_t mask);

    bool setChannelMasks(int name,
                         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);

    static void track__genericResample(track_t *t, int32_t *out, size_t numFrames, int32_t *temp,
                                       int32_t *aux);
    static void track__nop(track_t *t, int32_t *out, size_t numFrames, int32_t *temp, int32_t *aux);
    static void track__16BitsStereo(track_t *t, int32_t *out, size_t numFrames, int32_t *temp,
                                    int32_t *aux);
    static void track__16BitsMono(track_t *t, int32_t *out, size_t numFrames, int32_t *temp,
                                  int32_t *aux);
    static void volumeRampStereo(track_t *t, int32_t *out, size_t frameCount, int32_t *temp,
                                 int32_t *aux);
    static void volumeStereo(track_t *t, int32_t *out, size_t frameCount, int32_t *temp,
                             int32_t *aux);

    static void process__validate(state_t *state, int64_t pts);
    static void process__nop(state_t *state, int64_t pts);
    static void process__genericNoResampling(state_t *state, int64_t pts);
    static void process__genericResampling(state_t *state, int64_t pts);
    static void process__OneTrack16BitsStereoNoResampling(state_t *state,
                                                          int64_t pts);

    static int64_t calculateOutputPTS(const track_t &t, int64_t basePTS,
                                      int outputFrameIndex);

    static uint64_t sLocalTimeFreq;
    static pthread_once_t sOnceControl;
    static void sInitRoutine();

    /* multi-format volume mixing function (calls template functions
     * in AudioMixerOps.h).  The template parameters are as follows:
     *
     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
     *   USEFLOATVOL (set to true if float volume is used)
     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
     *   TO: int32_t (Q4.27) or float
     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
     *   TA: int32_t (Q4.27)
     */
    template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
              typename TO, typename TI, typename TA>
    static void volumeMix(TO *out, size_t outFrames,
                          const TI *in, TA *aux, bool ramp, AudioMixer::track_t *t);

    // multi-format process hooks
    template <int MIXTYPE, typename TO, typename TI, typename TA>
    static void process_NoResampleOneTrack(state_t *state, int64_t pts);

    // multi-format track hooks
    template <int MIXTYPE, typename TO, typename TI, typename TA>
    static void track__Resample(track_t *t, TO *out, size_t frameCount,
                                TO *temp __unused, TA *aux);
    template <int MIXTYPE, typename TO, typename TI, typename TA>
    static void track__NoResample(track_t *t, TO *out, size_t frameCount,
                                  TO *temp __unused, TA *aux);

    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
                                   void *in, audio_format_t mixerInFormat, size_t sampleCount);

    // hook types
    enum {
        PROCESSTYPE_NORESAMPLEONETRACK,
    };
    enum {
        TRACKTYPE_NOP,
        TRACKTYPE_RESAMPLE,
        TRACKTYPE_NORESAMPLE,
        TRACKTYPE_NORESAMPLEMONO,
    };

    // functions for determining the proper process and track hooks.
    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
                                         audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
    static hook_t getTrackHook(int trackType, uint32_t channelCount,
                               audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
};

// ----------------------------------------------------------------------------
} // namespace cc
